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Side by Side Diff: webrtc/media/engine/fakewebrtcvoiceengine.h

Issue 2737633002: Remove VoEHardware interface usage. (Closed)
Patch Set: comments + compile issue Created 3 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2010 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2010 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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52 52
53 #define WEBRTC_BOOL_STUB(method, args) \ 53 #define WEBRTC_BOOL_STUB(method, args) \
54 bool method args override { return true; } 54 bool method args override { return true; }
55 55
56 #define WEBRTC_VOID_STUB(method, args) \ 56 #define WEBRTC_VOID_STUB(method, args) \
57 void method args override {} 57 void method args override {}
58 58
59 #define WEBRTC_FUNC(method, args) int method args override 59 #define WEBRTC_FUNC(method, args) int method args override
60 60
61 class FakeWebRtcVoiceEngine 61 class FakeWebRtcVoiceEngine
62 : public webrtc::VoEBase, public webrtc::VoECodec, 62 : public webrtc::VoEBase, public webrtc::VoECodec {
63 public webrtc::VoEHardware {
64 public: 63 public:
65 struct Channel { 64 struct Channel {
66 std::vector<webrtc::CodecInst> recv_codecs; 65 std::vector<webrtc::CodecInst> recv_codecs;
67 size_t neteq_capacity = 0; 66 size_t neteq_capacity = 0;
68 bool neteq_fast_accelerate = false; 67 bool neteq_fast_accelerate = false;
69 }; 68 };
70 69
71 explicit FakeWebRtcVoiceEngine(webrtc::AudioProcessing* apm, 70 explicit FakeWebRtcVoiceEngine(webrtc::AudioProcessing* apm,
72 webrtc::voe::TransmitMixer* transmit_mixer) 71 webrtc::voe::TransmitMixer* transmit_mixer)
73 : apm_(apm), transmit_mixer_(transmit_mixer) { 72 : apm_(apm), transmit_mixer_(transmit_mixer) {
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196 } 195 }
197 WEBRTC_STUB(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode, 196 WEBRTC_STUB(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode,
198 bool disableDTX)); 197 bool disableDTX));
199 WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled, 198 WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
200 webrtc::VadModes& mode, bool& disabledDTX)); 199 webrtc::VadModes& mode, bool& disabledDTX));
201 WEBRTC_STUB(SetFECStatus, (int channel, bool enable)); 200 WEBRTC_STUB(SetFECStatus, (int channel, bool enable));
202 WEBRTC_STUB(GetFECStatus, (int channel, bool& enable)); 201 WEBRTC_STUB(GetFECStatus, (int channel, bool& enable));
203 WEBRTC_STUB(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)); 202 WEBRTC_STUB(SetOpusMaxPlaybackRate, (int channel, int frequency_hz));
204 WEBRTC_STUB(SetOpusDtx, (int channel, bool enable_dtx)); 203 WEBRTC_STUB(SetOpusDtx, (int channel, bool enable_dtx));
205 204
206 // webrtc::VoEHardware
207 WEBRTC_STUB(GetNumOfRecordingDevices, (int& num));
208 WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num));
209 WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid));
210 WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid));
211 WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel));
212 WEBRTC_STUB(SetPlayoutDevice, (int));
213 WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
214 WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
215 WEBRTC_STUB(SetRecordingSampleRate, (unsigned int samples_per_sec));
216 WEBRTC_STUB_CONST(RecordingSampleRate, (unsigned int* samples_per_sec));
217 WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec));
218 WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec));
219 WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
220 bool BuiltInAECIsAvailable() const override { return false; }
221 WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
222 bool BuiltInAGCIsAvailable() const override { return false; }
223 WEBRTC_STUB(EnableBuiltInNS, (bool enable));
224 bool BuiltInNSIsAvailable() const override { return false; }
225
226 size_t GetNetEqCapacity() const { 205 size_t GetNetEqCapacity() const {
227 auto ch = channels_.find(last_channel_); 206 auto ch = channels_.find(last_channel_);
228 RTC_DCHECK(ch != channels_.end()); 207 RTC_DCHECK(ch != channels_.end());
229 return ch->second->neteq_capacity; 208 return ch->second->neteq_capacity;
230 } 209 }
231 bool GetNetEqFastAccelerate() const { 210 bool GetNetEqFastAccelerate() const {
232 auto ch = channels_.find(last_channel_); 211 auto ch = channels_.find(last_channel_);
233 RTC_CHECK(ch != channels_.end()); 212 RTC_CHECK(ch != channels_.end());
234 return ch->second->neteq_fast_accelerate; 213 return ch->second->neteq_fast_accelerate;
235 } 214 }
236 215
237 private: 216 private:
238 bool inited_ = false; 217 bool inited_ = false;
239 int last_channel_ = -1; 218 int last_channel_ = -1;
240 std::map<int, Channel*> channels_; 219 std::map<int, Channel*> channels_;
241 bool fail_create_channel_ = false; 220 bool fail_create_channel_ = false;
242 webrtc::AudioProcessing* apm_ = nullptr; 221 webrtc::AudioProcessing* apm_ = nullptr;
243 webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr; 222 webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
244 223
245 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FakeWebRtcVoiceEngine); 224 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FakeWebRtcVoiceEngine);
246 }; 225 };
247 226
248 } // namespace cricket 227 } // namespace cricket
249 228
250 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 229 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
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